Add functions to interact with ASan and MSan, and some sample uses

webrtc/modules/audio_coding/codecs/audio_decoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 442ddc1e4b86b081653da168b363bf26e6bc9683..8d4a2bc175ba624bfc518adaf4fcebe9eea8d795 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -12,7 +12,9 @@
 
 #include <assert.h>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/sanitizer.h"
 #include "webrtc/base/trace_event.h"
 
 namespace webrtc {
@@ -21,6 +23,7 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
                          int sample_rate_hz, size_t max_decoded_bytes,
                          int16_t* decoded, SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
   int duration = PacketDuration(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@@ -34,6 +37,7 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
                                   int sample_rate_hz, size_t max_decoded_bytes,
                                   int16_t* decoded, SpeechType* speech_type) {
   TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
   int duration = PacketDurationRedundant(encoded, encoded_len);
   if (duration >= 0 &&
       duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {